.\" $NetBSD: audio.4,v 1.79.2.4 2019/09/02 16:39:20 martin Exp $ .\" .\" Copyright (c) 1996 The NetBSD Foundation, Inc. .\" All rights reserved. .\" .\" This code is derived from software contributed to The NetBSD Foundation .\" by John T. Kohl. .\" .\" Redistribution and use in source and binary forms, with or without .\" modification, are permitted provided that the following conditions .\" are met: .\" 1. Redistributions of source code must retain the above copyright .\" notice, this list of conditions and the following disclaimer. .\" 2. Redistributions in binary form must reproduce the above copyright .\" notice, this list of conditions and the following disclaimer in the .\" documentation and/or other materials provided with the distribution. .\" .\" THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS .\" ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED .\" TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR .\" PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS .\" BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR .\" CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF .\" SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS .\" INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN .\" CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) .\" ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE .\" POSSIBILITY OF SUCH DAMAGE. .\" .Dd May 15, 2018 .Dt AUDIO 4 .Os .Sh NAME .Nm audio .Nd device-independent audio driver layer .Sh SYNOPSIS .In sys/audioio.h .Sh DESCRIPTION The .Nm driver provides support for various audio peripherals. It provides a uniform programming interface layer above different underlying audio hardware drivers. The audio layer provides full-duplex operation if the underlying hardware configuration supports it. .Pp There are four device files available for audio operation: .Pa /dev/audio , .Pa /dev/sound , .Pa /dev/audioctl , and .Pa /dev/mixer . .Pp .Pa /dev/audio and .Pa /dev/sound are used for recording or playback of digital samples. .Pp .Pa /dev/mixer is used to manipulate volume, recording source, or other audio mixer functions. .Pp .Pa /dev/audioctl accepts the same .Xr ioctl 2 operations as .Pa /dev/sound , but no other operations. .Pp .Pa /dev/sound and .Pa /dev/audio can be opened at .Em any time and audio sources of different precision and playback parameters i.e frequency will be mixed and played back simultaneously. .Pp .Pa /dev/audioctl can be used to manipulate the audio device while it is in use. .Sh SAMPLING DEVICES When .Pa /dev/audio is opened, it automatically directs the underlying driver to manipulate monaural 8-bit mu-law samples. In addition, if it is opened read-only (write-only) the device is set to half-duplex record (play) mode with recording (playing) unpaused and playing (recording) paused. When .Pa /dev/sound is opened, it maintains the previous audio sample mode and record/playback mode most recently set on .Pa /dev/sound by any open channel. In all other respects .Pa /dev/audio and .Pa /dev/sound are identical. .Sh VIRTUAL CHANNELS Any process may open a sampling device at a given time. Any number of devices per process and file descriptors may be shared between processes. .Pp Virtual channels are converted to a common format, signed linear encoding, frequency channels and precision. These can be modified to taste by the following .Xr sysctl 8 variables. .Bl -tag -width -compact .It hw.driverN.precision .It hw.driverN.frequency .It hw.driverN.channels .It hw.driverN.latency .It hw.driverN.multiuser .El .Pp Where .Em driverN corresponds to the underlying audio device driver and device number. e.g In the case of an hdafg supported device the variables would be: hw.hdafg0.channels, hw.hdafg0.precision, hw.hdafg0.frequency. .Pp For best results, values close to the underlying hardware should be chosen. These variables may only be changed when the sampling device is not in use. .Pp The hw.driverN.latency .Xr sysctl 8 variable controls the latency of the in-kernel mixer by varying the hardware blocksize. It accepts a value in milliseconds(ms), fractional values are not allowed. A value of zero will default to 150ms. .Pp If a static blocksize is enforced by the underlying hardware driver this value cannot be changed. .Pp For audio applications that do not specify a preferred blocksize when configuring the audio device, this will be the latency these applications have. .Pp For audio applications that .Xr mmap 2 the audio device for play back the resultant latency is a third (1/3) of the value of the hw.driverN.latency variable. .Pp The hw.driverN.multiuser .Xr sysctl 8 variable determines if multiple users are allowed to access the sampling device. .Pp By default it is set to false. This means that the sampling device may be only used by .Em one user at a time. Other users (except root) attempting to open the sampling device will be denied. .Pp If set to true, all users may access the sampling device at any time. .Pp Each virtual channel has a corresponding mixer: .Bl -tag -width -compact .It vchan.dacN Output volume .It vchan.micN Recording volume .El .Pp Where N is the virtual channel number. e.g vchan.dac0 controlling playback volume and vchan.mic0 controlling recording volume for the first virtual channel. .Pp On a half-duplex device, writes while recording is in progress will be immediately discarded. Similarly, reads while playback is in progress will be filled with silence but delayed to return at the current sampling rate. If both playback and recording are requested on a half-duplex device, playback mode takes precedence and recordings will get silence. .Pp On a full-duplex device, reads and writes may operate concurrently without interference. If a full-duplex capable audio device is opened for both reading and writing it will start in half-duplex play mode; full-duplex mode has to be set explicitly. .Pp On either type of device, if the playback mode is paused then silence is played instead of the provided samples, and if recording is paused then the process blocks in .Xr read 2 until recording is unpaused. .Pp If a writing process does not call .Xr write 2 frequently enough to provide samples at the pace the hardware consumes them silence is inserted. If the .Dv AUMODE_PLAY_ALL mode is not set the writing process must provide enough data via subsequent write calls to .Dq catch up in time to the current audio block before any more process-provided samples will be played. If a reading process does not call .Xr read 2 frequently enough, it will simply miss samples. .Pp The audio device is normally accessed with .Xr read 2 or .Xr write 2 calls, but it can also be mapped into user memory with .Xr mmap 2 Once the device has been mapped it can no longer be accessed by read or write; all access is by reading and writing to the mapped memory. The device appears as a block of memory of size .Va buffersize (as available via .Dv AUDIO_GETINFO or .Dv AUDIO_GETBUFINFO ) . The device driver will continuously move data from this buffer from/to the audio hardware, wrapping around at the end of the buffer. To find out where the hardware is currently accessing data in the buffer the .Dv AUDIO_GETIOFFS and .Dv AUDIO_GETOOFFS calls can be used. The playing and recording buffers are distinct and must be mapped separately if both are to be used. Only encodings that are not emulated (i.e. where .Dv AUDIO_ENCODINGFLAG_EMULATED is not set) work properly for a mapped device. .Pp The audio device, like most devices, can be used in .Va select , can be set in non-blocking mode and can be set (with a .Dv FIOASYNC ioctl) to send a .Dv SIGIO when I/O is possible. The mixer device can be set to generate a .Dv SIGIO whenever a mixer value is changed. .Pp The following .Xr ioctl 2 commands are supported on the sample devices: .Bl -tag -width indent .It Dv AUDIO_GETCHAN (int) This command will return the audio channel in use. .It Dv AUDIO_SETCHAN (int) This command will select the audio channel for subsequent ioctl calls. .It Dv AUDIO_FLUSH This command stops all playback and recording, clears all queued buffers, resets error counters, and restarts recording and playback as appropriate for the current sampling mode. .It Dv AUDIO_RERROR (int) This command fetches the count of dropped input samples into its integer argument. There is no information regarding when in the sample stream they were dropped. .It Dv AUDIO_WSEEK (u_long) This command fetches the count of samples that are queued ahead of the first sample in the most recent sample block written into its integer argument. .It Dv AUDIO_DRAIN This command suspends the calling process until all queued playback samples have been played by the hardware. .It Dv AUDIO_GETDEV (audio_device_t) This command fetches the current hardware device information into the audio_device_t argument. .Bd -literal typedef struct audio_device { char name[MAX_AUDIO_DEV_LEN]; char version[MAX_AUDIO_DEV_LEN]; char config[MAX_AUDIO_DEV_LEN]; } audio_device_t; .Ed .It Dv AUDIO_GETFD (int) The command returns the current setting of the full duplex mode. .It Dv AUDIO_GETENC (audio_encoding_t) This command is used iteratively to fetch sample encoding names and format_ids into the input/output audio_encoding_t argument. .Bd -literal typedef struct audio_encoding { int index; /* input: nth encoding */ char name[MAX_AUDIO_DEV_LEN]; /* name of encoding */ int encoding; /* value for encoding parameter */ int precision; /* value for precision parameter */ int flags; #define AUDIO_ENCODINGFLAG_EMULATED 1 /* software emulation mode */ } audio_encoding_t; .Ed .Pp To query all the supported encodings, start with an index field of 0 and continue with successive encodings (1, 2, ...) until the command returns an error. .It Dv AUDIO_SETFD (int) This command sets the device into full-duplex operation if its integer argument has a non-zero value, or into half-duplex operation if it contains a zero value. If the device does not support full-duplex operation, attempting to set full-duplex mode returns an error. .It Dv AUDIO_GETPROPS (int) This command gets a bit set of hardware properties. If the hardware has a certain property the corresponding bit is set, otherwise it is not. The properties can have the following values: .Pp .Bl -tag -width AUDIO_PROP_INDEPENDENT -compact .It Dv AUDIO_PROP_FULLDUPLEX the device admits full duplex operation. .It Dv AUDIO_PROP_MMAP the device can be used with .Xr mmap 2 . .It Dv AUDIO_PROP_INDEPENDENT the device can set the playing and recording encoding parameters independently. .It Dv AUDIO_PROP_PLAYBACK the device is capable of audio playback. .It Dv AUDIO_PROP_CAPTURE the device is capable of audio capture. .El .It Dv AUDIO_GETIOFFS (audio_offset_t) .It Dv AUDIO_GETOOFFS (audio_offset_t) This command fetches the current offset in the input(output) buffer where the audio hardware's DMA engine will be putting(getting) data. It mostly useful when the device buffer is available in user space via the .Xr mmap 2 call. The information is returned in the audio_offset structure. .Bd -literal typedef struct audio_offset { u_int samples; /* Total number of bytes transferred */ u_int deltablks; /* Blocks transferred since last checked */ u_int offset; /* Physical transfer offset in buffer */ } audio_offset_t; .Ed .It Dv AUDIO_GETINFO (audio_info_t) .It Dv AUDIO_GETBUFINFO (audio_info_t) .It Dv AUDIO_SETINFO (audio_info_t) Get or set audio information as encoded in the audio_info structure. .Bd -literal typedef struct audio_info { struct audio_prinfo play; /* info for play (output) side */ struct audio_prinfo record; /* info for record (input) side */ u_int monitor_gain; /* input to output mix */ /* BSD extensions */ u_int blocksize; /* H/W read/write block size */ u_int hiwat; /* output high water mark */ u_int lowat; /* output low water mark */ u_int _ispare1; u_int mode; /* current device mode */ #define AUMODE_PLAY 0x01 #define AUMODE_RECORD 0x02 #define AUMODE_PLAY_ALL 0x04 /* do not do real-time correction */ } audio_info_t; .Ed .Pp When setting the current state with .Dv AUDIO_SETINFO , the audio_info structure should first be initialized with .Dv AUDIO_INITINFO Po &info Pc and then the particular values to be changed should be set. This allows the audio driver to only set those things that you wish to change and eliminates the need to query the device with .Dv AUDIO_GETINFO or .Dv AUDIO_GETBUFINFO first. .Pp The .Va mode field should be set to .Dv AUMODE_PLAY , .Dv AUMODE_RECORD , .Dv AUMODE_PLAY_ALL , or a bitwise OR combination of the three. Only full-duplex audio devices support simultaneous record and playback. .Pp .Va hiwat and .Va lowat are used to control write behavior. Writes to the audio devices will queue up blocks until the high-water mark is reached, at which point any more write calls will block until the queue is drained to the low-water mark. .Va hiwat and .Va lowat set those high- and low-water marks (in audio blocks). The default for .Va hiwat is the maximum value and for .Va lowat 75 % of .Va hiwat . .Pp .Va blocksize sets the current audio blocksize. The generic audio driver layer and the hardware driver have the opportunity to adjust this block size to get it within implementation-required limits. Upon return from an .Dv AUDIO_SETINFO call, the actual blocksize set is returned in this field. Normally the .Va blocksize is calculated to correspond to 50ms of sound and it is recalculated when the encoding parameter changes, but if the .Va blocksize is set explicitly this value becomes sticky, i.e., it remains even when the encoding is changed. The stickiness can be cleared by reopening the device or setting the .Va blocksize to 0. .Bd -literal struct audio_prinfo { u_int sample_rate; /* sample rate in samples/s */ u_int channels; /* number of channels, usually 1 or 2 */ u_int precision; /* number of bits/sample */ u_int encoding; /* data encoding (AUDIO_ENCODING_* below) */ u_int gain; /* volume level */ u_int port; /* selected I/O port */ u_long seek; /* BSD extension */ u_int avail_ports; /* available I/O ports */ u_int buffer_size; /* total size audio buffer */ u_int _ispare[1]; /* Current state of device: */ u_int samples; /* number of samples */ u_int eof; /* End Of File (zero-size writes) counter */ u_char pause; /* non-zero if paused, zero to resume */ u_char error; /* non-zero if underflow/overflow occurred */ u_char waiting; /* non-zero if another process hangs in open */ u_char balance; /* stereo channel balance */ u_char cspare[2]; u_char open; /* non-zero if currently open */ u_char active; /* non-zero if I/O is currently active */ }; .Ed .Pp Note: many hardware audio drivers require identical playback and recording sample rates, sample encodings, and channel counts. The playing information is always set last and will prevail on such hardware. If the hardware can handle different settings the .Dv AUDIO_PROP_INDEPENDENT property is set. .Pp The encoding parameter can have the following values: .Pp .Bl -tag -width AUDIO_ENCODING_SLINEAR_BE -compact .It Dv AUDIO_ENCODING_ULAW mu-law encoding, 8 bits/sample .It Dv AUDIO_ENCODING_ALAW A-law encoding, 8 bits/sample .It Dv AUDIO_ENCODING_SLINEAR two's complement signed linear encoding with the platform byte order .It Dv AUDIO_ENCODING_ULINEAR unsigned linear encoding with the platform byte order .It Dv AUDIO_ENCODING_ADPCM ADPCM encoding, 8 bits/sample .It Dv AUDIO_ENCODING_SLINEAR_LE two's complement signed linear encoding with little endian byte order .It Dv AUDIO_ENCODING_SLINEAR_BE two's complement signed linear encoding with big endian byte order .It Dv AUDIO_ENCODING_ULINEAR_LE unsigned linear encoding with little endian byte order .It Dv AUDIO_ENCODING_ULINEAR_BE unsigned linear encoding with big endian byte order .It Dv AUDIO_ENCODING_AC3 Dolby Digital AC3 .El .Pp The .Va gain , .Va port and .Va balance settings provide simple shortcuts to the richer mixer interface described below and are not obtained by .Dv AUDIO_GETBUFINFO . The gain should be in the range .Bq Dv AUDIO_MIN_GAIN , Dv AUDIO_MAX_GAIN and the balance in the range .Bq Dv AUDIO_LEFT_BALANCE , Dv AUDIO_RIGHT_BALANCE with the normal setting at .Dv AUDIO_MID_BALANCE . .Pp The input port should be a combination of: .Pp .Bl -tag -width AUDIO_MICROPHONE -compact .It Dv AUDIO_MICROPHONE to select microphone input. .It Dv AUDIO_LINE_IN to select line input. .It Dv AUDIO_CD to select CD input. .El .Pp The output port should be a combination of: .Pp .Bl -tag -width AUDIO_HEADPHONE -compact .It Dv AUDIO_SPEAKER to select speaker output. .It Dv AUDIO_HEADPHONE to select headphone output. .It Dv AUDIO_LINE_OUT to select line output. .El .Pp The available ports can be found in .Va avail_ports .Dv ( AUDIO_GETBUFINFO only). .Pp .Va buffer_size is the total size of the audio buffer. The buffer size divided by the .Va blocksize gives the maximum value for .Va hiwat . Currently the .Va buffer_size can only be read and not set. .Pp The .Va seek and .Va samples fields are only used by .Dv AUDIO_GETINFO and .Dv AUDIO_GETBUFINFO . .Va seek represents the count of samples pending; .Va samples represents the total number of bytes recorded or played, less those that were dropped due to inadequate consumption/production rates. .Pp .Va pause returns the current pause/unpause state for recording or playback. For .Dv AUDIO_SETINFO , if the pause value is specified it will either pause or unpause the particular direction. .El .Sh MIXER DEVICE The mixer device, .Pa /dev/mixer , may be manipulated with .Xr ioctl 2 but does not support .Xr read 2 or .Xr write 2 . It supports the following .Xr ioctl 2 commands: .Bl -tag -width indent .It Dv AUDIO_GETDEV (audio_device_t) This command is the same as described above for the sampling devices. .It Dv AUDIO_MIXER_READ (mixer_ctrl_t) .It Dv AUDIO_MIXER_WRITE (mixer_ctrl_t) These commands read the current mixer state or set new mixer state for the specified device .Va dev . .Va type identifies which type of value is supplied in the .Va mixer_ctrl_t argument. .Bd -literal #define AUDIO_MIXER_CLASS 0 #define AUDIO_MIXER_ENUM 1 #define AUDIO_MIXER_SET 2 #define AUDIO_MIXER_VALUE 3 typedef struct mixer_ctrl { int dev; /* input: nth device */ int type; union { int ord; /* enum */ int mask; /* set */ mixer_level_t value; /* value */ } un; } mixer_ctrl_t; #define AUDIO_MIN_GAIN 0 #define AUDIO_MAX_GAIN 255 typedef struct mixer_level { int num_channels; u_char level[8]; /* [num_channels] */ } mixer_level_t; #define AUDIO_MIXER_LEVEL_MONO 0 #define AUDIO_MIXER_LEVEL_LEFT 0 #define AUDIO_MIXER_LEVEL_RIGHT 1 .Ed .Pp For a mixer value, the .Va value field specifies both the number of channels and the values for each channel. If the channel count does not match the current channel count, the attempt to change the setting may fail (depending on the hardware device driver implementation). For an enumeration value, the .Va ord field should be set to one of the possible values as returned by a prior .Dv AUDIO_MIXER_DEVINFO command. The type .Dv AUDIO_MIXER_CLASS is only used for classifying particular mixer device types and is not used for .Dv AUDIO_MIXER_READ or .Dv AUDIO_MIXER_WRITE . .It Dv AUDIO_MIXER_DEVINFO (mixer_devinfo_t) This command is used iteratively to fetch audio mixer device information into the input/output mixer_devinfo_t argument. To query all the supported devices, start with an index field of 0 and continue with successive devices (1, 2, ...) until the command returns an error. .Bd -literal typedef struct mixer_devinfo { int index; /* input: nth mixer device */ audio_mixer_name_t label; int type; int mixer_class; int next, prev; #define AUDIO_MIXER_LAST -1 union { struct audio_mixer_enum { int num_mem; struct { audio_mixer_name_t label; int ord; } member[32]; } e; struct audio_mixer_set { int num_mem; struct { audio_mixer_name_t label; int mask; } member[32]; } s; struct audio_mixer_value { audio_mixer_name_t units; int num_channels; int delta; } v; } un; } mixer_devinfo_t; .Ed .Pp The .Va label field identifies the name of this particular mixer control. The .Va index field may be used as the .Va dev field in .Dv AUDIO_MIXER_READ and .Dv AUDIO_MIXER_WRITE commands. The .Va type field identifies the type of this mixer control. Enumeration types are typically used for on/off style controls (e.g. a mute control) or for input/output device selection (e.g. select recording input source from CD, line in, or microphone). Set types are similar to enumeration types but any combination of the mask bits can be used. .Pp The .Va mixer_class field identifies what class of control this is. The .Pq arbitrary value set by the hardware driver may be determined by examining the .Va mixer_class field of the class itself, a mixer of type .Dv AUDIO_MIXER_CLASS . For example, a mixer controlling the input gain on the line in circuit would have a .Va mixer_class that matches an input class device with the name .Dq inputs .Dv ( AudioCinputs ) , and would have a .Va label of .Dq line .Dv ( AudioNline ) . Mixer controls which control audio circuitry for a particular audio source (e.g. line-in, CD in, DAC output) are collected under the input class, while those which control all audio sources (e.g. master volume, equalization controls) are under the output class. Hardware devices capable of recording typically also have a record class, for controls that only affect recording, and also a monitor class. .Pp The .Va next and .Va prev may be used by the hardware device driver to provide hints for the next and previous devices in a related set (for example, the line in level control would have the line in mute as its .Dq next value). If there is no relevant next or previous value, .Dv AUDIO_MIXER_LAST is specified. .Pp For .Dv AUDIO_MIXER_ENUM mixer control types, the enumeration values and their corresponding names are filled in. For example, a mute control would return appropriate values paired with .Dv AudioNon and .Dv AudioNoff . For .Dv AUDIO_MIXER_VALUE and .Dv AUDIO_MIXER_SET mixer control types, the channel count is returned; the units name specifies what the level controls (typical values are .Dv AudioNvolume , .Dv AudioNtreble , .Dv AudioNbass ) . .\" For AUDIO_MIXER_SET mixer control types, what is what? .El .Pp By convention, all the mixer devices can be distinguished from other mixer controls because they use a name from one of the .Dv AudioC* string values. .Sh FILES .Bl -tag -width /dev/audioctl -compact .It Pa /dev/audio .It Pa /dev/audioctl .It Pa /dev/sound .It Pa /dev/mixer .El .Sh SEE ALSO .Xr audioctl 1 , .Xr mixerctl 1 , .Xr ioctl 2 , .Xr ossaudio 3 , .Xr midi 4 , .Xr radio 4 , .Xr sysctl 8 .Ss ISA bus .Xr aria 4 , .Xr ess 4 , .Xr gus 4 , .Xr guspnp 4 , .Xr pas 4 , .Xr sb 4 , .Xr wss 4 , .Xr ym 4 .Ss PCI bus .Xr auacer 4 , .Xr auich 4 , .Xr auixp 4 , .Xr autri 4 , .Xr auvia 4 , .Xr azalia 4 , .Xr clcs 4 , .Xr clct 4 , .Xr cmpci 4 , .Xr eap 4 , .Xr emuxki 4 , .Xr esa 4 , .Xr esm 4 , .Xr eso 4 , .Xr fms 4 , .Xr neo 4 , .Xr sv 4 , .Xr yds 4 .Ss TURBOchannel .Xr bba 4 .Ss USB .Xr uaudio 4 .Ss The NetBSD audio specification .Xr audio_system 9 .Sh HISTORY Support for virtual channels and mixing first appeared in .Nx 8.0 .